Conventionally, there has been a desire to faithfully convert an electric signal into a sound wave in a normal loudspeaker which does not perform electric signal processing. However, it is hard for an actual loudspeaker to perform faithful conversion due to limitations on its structure. For example, in a magnetic circuit constituting the loudspeaker, because of its structure, a magnetic flux density in a magnetic gap decreases as amplitude increases. Then, a force coefficient also decreases with the decrease of the magnetic flux density. The stiffness of a support system such as a damper, an edge, and the like changes according to the magnitude of the amplitude because of the structure of the support system. Due to these reasons, the amplitude of the loudspeaker is not necessarily proportional to the magnitude of the inputted electric signal, and there is a problem that non-linear distortion occurs.
As a method of removing the above non-linear distortion, conventionally, there has been proposed a method using electric signal processing such as feedforward processing, or the like. This processing method is a method in which polynomial approximation is performed on a parameter (a force coefficient according to a magnetic flux density, a stiffness of a support system, or the like) including a non-linear component of the loudspeaker and a filter coefficient is set so as to cancel non-linear distortion attributable to the parameter. An electric signal is inputted to the loudspeaker through a filter the filter coefficient of which is set, thereby removing the non-linear distortion. However, especially, the stiffness of the support system among the parameter changes hourly, and also ages. In other words, the value of the parameter changes over time. Thus, in the above feedforward processing, error between the preset value of the parameter and the actual value of the parameter becomes large over time, and there is a drawback that the above effect of distortion removal is significantly deteriorated.
For solving the above problem, in the feedforward processing, there has been proposed a method to adaptively update the parameter of the filter coefficient (e.g. refer to Patent Document 1). The following will describe this method with reference to FIG. 28. FIG. 28 is a block diagram showing a conventional loudspeaker device 9 which adaptively updates the parameter of the filter coefficient.
In FIG. 28, the conventional loudspeaker device 9 includes a control section 91, a parameter detector 92, and a loudspeaker 95. The parameter detector 92 includes an error circuit 93 and an update circuit 94. The error circuit 93 includes a filter (not shown), and calculates at the filter a pseudo vibration characteristic from a signal inputted from the control section 91. The error circuit 93 predictively calculates from the pseudo vibration characteristic a drive voltage which is applied to the loudspeaker 95. It is noted that the predicted drive voltage is equivalent to an impedance characteristic when the loudspeaker 95 is driven by a current. Then, the error circuit 93 produces an error signal e(t) by subtracting an actual drive voltage which is applied to the loudspeaker 95 from the predicted drive voltage. The error signal e(t) is inputted to the update circuit 94.
Based on the error signal e(t), the update circuit 94 calculates a parameter in the control section 91, which is to be updated. The parameter calculated by the update circuit 94 is reflected to the filter of the error circuit 93, and a gradient signal Sg is produced by the error circuit 93. The gradient signal Sg produced by the error circuit 93 is outputted to the update circuit 94 again. Thus, the update circuit 94 calculates a parameter using the above error signal e(t) and the gradient signal Sg so that the error signal e(t) becomes minimum. The parameter when the error signal e(t) becomes minimum is outputted as a power vector P to the control section 91, and the parameter in the control section 91 is updated. As described above, in the loudspeaker device 9 as shown in FIG. 28, the parameter is updated by the error circuit 93 and the update circuit 94 so that the parameter in the control section 91 corresponds to the parameter of the actual loudspeaker 95.
[Patent Document 1] Japanese Patent Laid-open Publication No. 11-46393